24 feb 2007 kl. 11.07 skrev Pavel Jezek:
Olle E Johansson wrote:
23 feb 2007 kl. 12.42 skrev Steve Davies:
Hi,
In older versions of asterisk I used to be able to use
incominglimit=1 to effectively disable call waiting on a specific
SIP channel (Where broken phones do not allow
22 feb 2007 kl. 23.40 skrev Philipp Kempgen:
Olle E Johansson wrote:
22 feb 2007 kl. 19.34 skrev Philipp Kempgen:
I thought it might be useful to be able to ask Asterisk for the
current SIP CSeq through the Manager API in order to send your
own SIP messages during a call outside
23 feb 2007 kl. 06.52 skrev Yuan LIU:
Quite a few documents, including voip-info, make reference to this
term. (e.g., First, You need trunk version of Asterisk.) But I
can't seem to find anything that defines this. In SVN, trunk
simply refers to the main body of code. Can someone
21 feb 2007 kl. 15.50 skrev James Fromm:
Anybody seen this behavior?
To determine if it's my config or a bug, could I trouble someone
running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP
interface as a test? After a few hours a 'sip show inuse' should
indicate the
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command? Thanks
for pointers.
None.
What do you want to do with SIP INFO?
/O
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22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to another via SIP and on
the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from
xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
Look at:
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to another via SIP and on
the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from
xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
These messages also
22 feb 2007 kl. 10.05 skrev Gordon Henderson:
On Thu, 22 Feb 2007, Rilawich Ango wrote:
Hi all,
I have read many forums and discussion groups talking about fax
support in asterisk. Some of them conclude that asterisk doesn't
support fax. However, some of them conclude that there is no
22 feb 2007 kl. 11.19 skrev Torbjörn Abrahamsson:
Well, but isn't lines that begin with -- on the same verbosity
level? So lowering the verbosity would in this case mean that you
also stop displaying the dialplan execution steps. I have a similar
problem regarding the -- SIP Seeding peer
21 feb 2007 kl. 12.54 skrev Steve Langstaff:
Hi All.
This is on Asterisk 1.2.13
I place a call between 2 SIP phones (with canreinvite=yes,
qualify=yes).
I reset the phones (so they don't have time to say BYE).
Asterisk seems to think that the call is still ongoing. This persists
until I
22 feb 2007 kl. 11.36 skrev Eric Bishop:
I do need MWI notifcation, just not on this particulary trunk. Is
there anyway to to turn off MWI on a particular trunk or can it
only be done globally?
You enable it per device in sip.conf - that's the only way.
/O
22 feb 2007 kl. 11.54 skrev Torbjörn Abrahamsson:
Agreed, but your response to the OP said to lower the verbosity,
and I commented that it might not be possible, due to then seeing
no dialplan execution... :)
Well if you want that level of detail during the execution, these
error messages
22 feb 2007 kl. 12.20 skrev Steve Langstaff:
Are the RTP timers applicable with canreinvite=yes ?
how could we possibly check RTP if the RTP doesn't touch or network
card at all?
The timers are only used when we have RTP streams going to us. If the
RTP stream
is redirected, it's up to
22 feb 2007 kl. 16.38 skrev Yehavi Bourvine +972-8-9489444:
Hello,
I am running version 1.4 with realtime support. I've set (for
Snom phones
300/320/360) a call limit of 1 (incominglimit and outgoinglimit
fields in the
database).
- When I used 1.4 SIP SHOW PEER show that it has a call
22 feb 2007 kl. 19.34 skrev Philipp Kempgen:
Yuan LIU wrote:
From: Olle E Johansson [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:36:45 +0100
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command?
Thanks for
pointers.
None.
What do you want
22 feb 2007 kl. 21.02 skrev Bill Gibbs:
Ray,
I have been playing with OpenPBX. My core servers are Asterisk so
I was playing around with their T38Gateway application. Long story
short - I can get the ATA (behind NAT) to talk T38 to the rxfax app
on an OpenPBX server but the gateway
22 feb 2007 kl. 22.30 skrev Sune Kloppenborg Jeppesen:
On Thursday 22 February 2007 22:24, Norbert Zawodsky wrote:
Hi everybody!
I've setup my dialplan so that if an extension dials *21*, that
extension is added/removed as a queue member to a queue. (State
toggled).
But it would be great
20 jan 2007 kl. 03.01 skrev Eric Bishop:
On inbound calls from my SIP provider I get multiple warnings as
follows:
WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid
host
Everything else works but these warnings are a pain and I don't
know what they are about
30 jan 2007 kl. 06.38 skrev Yuan LIU:
When Asterisk dials an IAX destination with no registration, it
very quickly comes to the conclusion that it can't make the call
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, IAX2/
[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called
the current status of implementing multiple parking lots.
Multiple parking lots in 1.4 is something we were hoping for and would
love to see happen.
I'd be happy to help with testing/debuging. Please feel free to
contact me.
On Fri, 26 Jan 2007 08:28:50 +0100, Olle E Johansson [EMAIL PROTECTED]
wrote
Thanks for this discussion! I've gotten a few ideas for better NAT
handling in chan_sip3.
The current way is implemented in so many installations, so it would
be hard to turn it
around, but in pineapple I can freely break backwards compatibility.
Let me think about it for a few days, then
26 jan 2007 kl. 16.31 skrev James Fromm:
Olle E Johansson wrote:
24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another
24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately hanging up will
clear the state.
The
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---
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* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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25 jan 2007 kl. 08.26 skrev Darryl Dunkin:
There is an SVN branch with this feature:
http://svn.digium.com/view/asterisk/team/oej/multiparking/
I had hope this would be a feature added to Asterisk 1.4, but fail to
see it on the changelog.
It wasn't approved due to some architecture issues.
22 jan 2007 kl. 07.38 skrev Thomas Deillon:
Hi all,
I read somewhere that asterisk v 1.4 can make Stun and SNMP.
I tried to find more information on these features but I didn’t
find any clues.
Someone find a way to use it?
There's a module called res_snmp that implements an SNMP agent
22 jan 2007 kl. 16.13 skrev dima:
Hello, everyone.
I'm reading about the asterisk new features. One is T.38 protocol
support. I used faxes before with asterisk 1.2 and everything was
working quite well. Could anyone explain what have changed in the way
faxes are handled.
Another thing is, in
23 jan 2007 kl. 10.42 skrev yusuf:
Hi,
I have a bunch of SIP phones(behind NAT) registering on my * box.
I want to find out when they register and de-register. I also want
to operate on it, so when they register/de-register, I want to
insert calldate into a mysql DB, etc.
Maybe
23 jan 2007 kl. 16.09 skrev Chris Bullock:
I'm running into an issue w/ Buddy status on Polycom IP650 phones
using
buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the
status on the
phones will stick in the busy status. I have noticed that I can
call that
extension the status
23 jan 2007 kl. 16.07 skrev Victor Toofic:
El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith
comentaba:
however, if I use sipp to test this, I get
[Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No
audio available on SIP/sipp-b7c274b0??
I suspect that's
31 okt 2006 kl. 01.37 skrev je .:
Thanks for the diagram. Is it possible to get a more detailed
diagram. I'm looking for something a little more technical. In
other words, where does Asterisk stand when inviting a user, when
hanging up, when canceling an invitation etc.. Does it go
proxy. If u1 hangs up,
we decide to hang up our call to u2,
but those are two different SIP dialogs.
Remember that Asterisk is a multiprotocol PBX. The connection to U2
might be using a different signalling
protocol, like ISDN PRI.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk
26 okt 2006 kl. 18.57 skrev Douglas Garstang:
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 26, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP v IAX2
On Thu, 2006-10-26 at 17:43
Thanks again
There is not video support at all in any of the H.323 channels
at this point in time. There are work in progress for the
H.323 channel based on OpenH323 (which includes video
support) but I don't know the state of that project.
Regards,
/olle
---
* Olle E. Johansson - [EMAIL
since the whole user/peer client/server concept does not
really match SIP.
In some cases, we're the SIP registrar/location server and in other
we're configured as the outbound proxy, even though we are not
a proxy.
I hope I did not add to the confusion by this confusing message.
/O
---
* Olle E
can't see how it could, but
I'd like to be surprised)
I would be *very* surprised if that worked!!!
I guess we have to implement the sip timer extension to be able to
solve that issue.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training
the details specs
for ISDN to SIP translation, some granularity is lost.
/O
---
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12 okt 2006 kl. 03.36 skrev Andrew Joakimsen:
What are your T.38 plans with this?
That's top secret... :-)
The T38 will be handled the same way as today - in passthrough mode -
until we have more T38 implementation code within the core. That's a bit
outside of the SIP scope.
/O :-)
12 okt 2006 kl. 15.51 skrev Jay R. Ashworth:
On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson wrote:
The new channel will have configurations for trunks, services and
phones. It will
Does that mean that it will make a distinction concerning the
difference in administrative span
training classes that provide
development funding
by attending the classes. Thanks!
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Next class: Stockholm, Sweden November 13-17 2006
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.
If you are
interested, please contact me quickly off list directly to my e-mail
address
[EMAIL PROTECTED]
Big thanks to Digium who has supported this work up to now.
Thanks for your support!
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* [EMAIL PROTECTED] - lot' of Asterisk activities, lots
and make sure they are the same.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk beachcamp: Bootcamp in Malaga, Spain - http://edvina.net/
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17 aug 2006 kl. 21.33 skrev Douglas Garstang:
Does realtime support include = yet?
Do you mean the realtime switch or static load of extensions.conf
from realtime?
If you mean the realtime switch, how would you suggest include=
should work?
Curious.
/Olle
17 aug 2006 kl. 22.03 skrev kjcsb:
How to I access the URI from an Invite:
INVITE sip:[EMAIL PROTECTED]
I want to set a variable to equal 5556678. The variable ${SIPURI}
returns the From URI.
The extension processed in the dialplan is the userinfo part of the
URI - the part before
the
(TO)}. What are the
others?
I would guess that you can check the RFC. Easier is to turn on SIP
debug and see the INVITE packet yourself and
check the headers that you have with your equipment.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training
, is something we have to find out from
the
RFCs - a task that is not always easy.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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16 aug 2006 kl. 07.26 skrev Dinesh Nair:
On 08/15/06 23:30 Michael J. Tubby B.Sc (Hons) G8TIC said the
following:
I suspect your problem is with the softphone implementation...
definitely, the SIP spec iianm says that UACs should play a ringing
tone when the 180 is received.
, please
always indicate
your version of Asterisk when you ask questions.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk Bootcamp Boston next week - open seats available!
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or
enough, I look forward
to your input and patches/suggestions.
/Olle
---
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* Asterisk Training http://edvina.net/training/
* Asterisk Beachcamp - Malaga, Spain! September 25-29
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.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* [EMAIL PROTECTED] VON Fall, Boston, Sept 11-14 - http://www.pulver.com/
asterisk/
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asterisk
Just a quick note that Edvina in cooperation with Digium is starting
the fall season of trainings again.
Coming trainings are:
* Asterisk Bootcamp, Boston - next week!
We still have a few seats available
* Asterisk Beachcamp, Malaga, Spain
A class in a beach hotel in beautiful Malaga on
[EMAIL PROTECTED] -
There will be a lot of Asterisk-related activities at Voice On the
Net FALL - Von - in Boston.
Apart from Digium booth (#819), there will be Asterisk presentations
as well as developer meetings.
For the [EMAIL PROTECTED] agenda, see http://www.pulver.com/asterisk/
-
The music on hold system was changed by Russell in trunk a few days
ago, since what you
describe is a bug reported long time ago by Terry Wilson.
In trunk/1.4, when you put someone on hold, the music played is
whatever is configured for the
device that puts someone else on hold. If you want
7 aug 2006 kl. 10.27 skrev Siqhamo Sifo:
Is there a way to offer Video conferrencing over Asterisk .
No.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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7 aug 2006 kl. 09.52 skrev Pavel Jezek:
maybe problem with expired nat translations at your firewall/nat
box? so asterisk can't contact your phone, through non existent
nat entry?
some sip debug on asterisk and etherreal packet dump on phone side
can help with solving this issue
in that it offers up its first choice
codec (only), and if asterisk supports that first choice, that's
what is used. Highly dependent on the sip phone manufacturers coding.
Yes. Asterisk has some interesting behaviour too, that we will have
to change to comply with recent standards.
/O
---
* Olle E
more than two
video points connected although there is a bounty for anyone that can
make something like meetme with video.
IAX2 also support video calls.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training
more flexibility. You will
be happy upgrading to 1.2 and even more happy upgrading to the
future 1.4 :-)
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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6 jul 2006 kl. 09.01 skrev unplug:
Hi all,
As we know, we can use dial plan to control the flow of the call
making process. However, I want to know how to control during the
registration process. Say, in ARA, during the registration process,
there is update sql statement in the user table.
Asterisk has some issues with the Nokia SIP client. I've started to
add some small
changes to svn trunk to support call hold with the Nokia, as well as
behave a bit
better in regards to ilbc encoding, even though that should still be
avoided.
I've had a lot of issues with the Nokia loosing
/tmp/asterisk-debug
The n will remove some ascii color codes. You might want to add a
g as
well to force core files if Asterisk crashes.
/o
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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for signaling?
SIP streams are signalling...
Sorry, I was talking about the media.
Have you tested the ACL features in sip.conf - accept/deny ?
Any pointers on these ACLs?
Check permit and deny in sip.conf.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net
20 jun 2006 kl. 08.51 skrev Mindaugas Kuprys:
Hi all,
I'm testing video phones with asterisk for the first time. Voice
calls goes fine. I have problems with video session. Advices needed!
here is asterisk log:
Jun 20 12:34:08 WARNING[16627]: chan_sip.c:3573 process_sdp:
Unknown SDP media
Want to become an Asterisk SIPmaster? Register for the Asterisk SIP
Master Class, taking place in Chicago, IL, USA
July 10-14 organized by Edvina in partnership with Digium. We're
developing this new training now, creating labs with
Asterisk and SIP express router, NAT traversals, realtime and
?
A semicolon in realtime separates multiple values, it is *not* used as a
comment. So you should fix your script.
regards,
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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20 jun 2006 kl. 14.28 skrev Mindaugas Kuprys:
Video started to work. Now intresting thing is that video size is
half reduced than calling directly from phone to phone. Phones:
Tatung tia-8800. I have attached sip messages. that else might be
important..? one of phones is behind nat.
a way to set an option
per device whether you want to play music or actually signal hold status
to the other end.
Right now, the other end will never know that it's on hold, it just
gets another
audio stream and merrily continues the call.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED
, will find it there.
And please don't forget to read the bug guidelines before you submit
the bug.
Regards,
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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Friends in the community,
I've received many mails saying I'll meet you at Astricon Europe.
The sad answer is no, you will not.
I have nothing to do with Astricon any more. After some arguments,
Steve decided that Astricon, trainings,
the business we had built together - everything
to
an exact extension.
Regards,
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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us test that code too.
Thank you all for working hard on this until I return online!
Regards,
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Next training and dCAP in Stockholm, Sweden, June 2006!
---
* Olle E Johansson - [EMAIL
this training:
Teacher: Olle E. Johansson, Asterisk developer and trainer.
Material: Training slides (over 300 pages), The Asterisk Quick
Reference Guide
Dates: June 12-16 (starting 10 AM Monday, ending noon friday)
Options: dCAP exam friday afternoon, June 16th
Price: 2.500 Euro (ex VAT). 200 Euro (ex VAT
Asterisk 1.4 for us.
We need your help, now. Download svn trunk and test in your environment!
On behalf of the community - thank you for testing!
SIP greetings!
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training
19 maj 2006 kl. 00.14 skrev Douglas Garstang:
Cool. Thanks. Now, I'm just wondering what SIP methods that will
work on? Need to peek into a REFER message from a phone.
I don't think that will work, but it is a cool idea since REFER
creates a new call and goes through the dial plan again.
.
Is Asterisk supposed to honor this field and retry with the proposed
minimum Expires: field? It looks like it doesn't, and I had to change
the default_expirey globally.
That's right, Asterisk is not aware of that header. Could I please see a
SIP debug trace of it?
/Olle
---
* Olle E. Johansson
.
Those may still be posted on this mailing list!
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Next class: Asterisk Bootcamp, Stockholm, June 12-16
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Just a quick reminder - now is the last chance to register for
MeetAsterisk in
Brussels on Thursday and London on Friday.
We have updated the web site with location information and will
keep registration open until tomorrow lunch.
http://www.meetasterisk.com
See you!
/Olle
Asterisk system
easier.
It's a big step forward and an important part of Asterisk 1.4.
Now, I have to learn the inner workings of this and adopt my branches
to it...
Always good to have something to do ;-)
Greetings from the Asterisk Developer Community!
/Olle
---
* Olle E. Johansson
.
Regards
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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25 apr 2006 kl. 15.34 skrev Wai Wu:
Hi,
What is the version number of the lastest stable release, and how
to get
it through CVS or wget? Thnx.
All of the information you look for is easily available on
http://www.asterisk.org
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
Friends,
Beginning next week, I will travel around Europe to teach Asterisk -
the one day Meet Asterisk training.
MeetAsterisk is organized by Edvina in cooperation with Digium and
Voop. In many places, local Asterisk
equipment resellers participate and show their equipment.
This is the
in
audio instead of sending an indication signal
Seems like there is a signalling problem so that Asterisk does not
get a ringing indication
from the device that we ring before moving to voicemail, or that
Broadvoice does not get
the ringing indication we send to them.
/O
---
* Olle E
no answer.
Sending DNS queries, not getting any response, kills Asterisk.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED] * MeetAsterisk European Tour
http://www.meetasterisk.com
* Asterisk Training http://edvina.net/training/
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11 apr 2006 kl. 16.05 skrev Brent Torrenga:
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before
the Cisco
79[46]0's start to ring.
If we were lucky enough
Please use the asterisk-biz mailing list for all commercial
offerings. Thank you.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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12 apr 2006 kl. 08.46 skrev Michael Strelnikov:
What caching DNS do you recommend?
Anyone you feel comfortable running.
/O
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12 apr 2006 kl. 09.08 skrev Cristian Draghici:
If DNS does not work on your local network, Asterisk will lock up.
Out of curiosity - the async implementation you mentioned in the other
thread - will it replace gethostbyname with something smarter or just
run things in a different thread
to Asterisk
on the same LAN, and Asterisk will
get out of the media path.
/Olle
---
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* Asterisk Training http://edvina.net/training/
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12 apr 2006 kl. 21.30 skrev David Gomillion:
If it's already been covered, please forgive the repetition. I
searched
Mantis, but couldn't come up with anything.
We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped
working on SP IP 300s and 600s. All of them.
I tried
errors on the console. What have I missed
that will let the
wellgate 3802 connect to asterisk?
You have no account named 1001. Your account names are
wellgate3802L1 and wellgate3802L2 in this configuration.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED] * Asterisk European Tour: http
Benoit Panizzon wrote:
Hi all
Hmm, often when my Asterisk tryes to register, it get's the answer back:
Got SIP response 302 Moved temporarely (and an IP).
But it looks like it's not respecting this redirection and tryes again and
again to register to the server configured in sip.conf
/demo-
moreinfo.gsm
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk European Tour: http://www.meetasterisk.com
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if you can make a bug reported in the bug tracker to show up
on your system and report that fact. Some bugs are hard to find, and
we need help finding out if a bug can be repeated or not.
Thank you for your assistance and understanding!
Asterisk bug marshals and developers
through
/O
---
* Olle E
5 apr 2006 kl. 16.40 skrev Jon Weisman:
Anyone know how I can get SIP T working w/ Asterisk?
Start with explaining your definition of SIP T then we can look
into it :-)
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training
not availble, please alert me and we'll try to fix it.
Thanks,
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk european tour: http://www.meetasterisk.com
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MeetAsterisk - a one day training arranged by Digium, Voop and
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equipment and meet Asterisk professionals!
Go to http://www.meetasterisk.com to learn more and register today!
29 mar 2006 kl. 01.03 skrev Douglas Garstang:
I made the post below earlier today. I'v since removed all NAT from
the equation and the problem still persists. Basically I am trying
to transfer a call. The transferring phone sends a REFER message to
asterisk with a call id that Asterisk
with Asterisk.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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